The present invention relates to voice processing systems, and in particular to the manner in which such voice processing systems may be attached to a switch via an ATM link.
Voice processing systems are well-known (see for example "Voice Processing" by Walt Teschner, published by Artech House), and are used to perform a variety of tasks, for example voice mail (also known as voice messaging), whereby callers who cannot reach their intended target can instead record a message for subsequent retrieval, and interactive voice response (IVR), in which callers interact with the voice processing system, typically by means of pressing DTMF keys, in order to obtain particular information.
Many other technologies are being incorporated into voice processing systems, such as voice recognition, text-to-speech, FAX, and so on. Furthermore, voice processing systems are now being considered for use as general media servers, for use for example in playing out video streams over the telephone network (or other appropriate network).
The simplest voice processing systems have as their input a conventional analog telephone line, in other words, they can plug into a socket in place of a normal telephone line. Multiple lines can be used if necessary, but this approach still has a very limited call capacity. Therefore more sophisticated voice processing systems have a digital trunk connection, either to a switch or directly into the digital telephone network. Such voice processing systems are often installed at the sites of customers who have relatively large volumes of incoming or outgoing telephone calls and therefore have their own switch. Thus the voice processing systems makes or receives telephone calls through the switch over one or more digital telephone lines.
Modern telephony signals are generally transmitted in digital form using standard 8-bit samples at a rate of 8 kHz, thereby requiring an overall bandwidth of 64 kilobits per second (kbps). The nature of the audio signals is such that an essentially logarithmic quantisation provides the best representation of the original analog input in the amplitude domain. Two quantisation laws in particular are widely used: A law in Europe and mu law in North America. These are defined in recommendation G.711 of the CCITT "Yellow Books" (now the ITU).
To allow large volumes of telephone traffic to be handled simply, individual signals are time multiplexed together for transmission over digital trunk lines. In North America, the standard form of trunk line is known as T1 (also referred to as DS1), and provides 24 simultaneous lines. These trunk lines can be used not only to carry the actual audio telephone signal, but also to provide a limited degree of signalling information, for example, to reserve a channel, to make a call on a channel, to transfer a call, and so on. In Europe, the standard form of trunk line is known as E1, and provides 32 simultaneous lines (of which 30 are for telephony channels, one for framing, and one for signalling).
Voice processing systems typically utilise such T1/E1 trunks as their telephony connection, either to a switch, or directly to the telephone network. This requires specialised telephony hardware, known as a line interface unit, capable of interfacing to the T1/E1 trunk, for performing multiplexing/demultiplexing, DTMF detection, signalling, and so on. The line interface unit can either be provided in the form of an adapter card within the voice processing system, or as a separate unit.
An example of a voice processing system which uses a separate unit to interface to a T1/E1 trunk is the DirectTalk/6000 system, available from IBM Corporation. This product is described in the manual "AIX DirectTalk/6000 General Information and Planning" (GC33-1720-00), also available from IBM, along with the other manuals listed therein.
FIG. 1 is an illustration of a DirectTalk/6000 system installation, including a switch 22 and DirectTalk/6000 voice processing system 60. The switch is connected by multiple T1 trunk lines 15 to the telephone network 10 (it is assumed that this installation is in North America, so that T1 lines are used, but it will be appreciated that the same configuration could be adopted also in Europe, using E1 trunk Lines).
The voice processing system 60 comprises two main components, a host processor 45, which for the DirectTalk/6000 system is a RISC System/6000 computer, and a set of one or more line interface units 25. The DirectTalk/6000 system supports a couple of different possible line interface units, but the one illustrated is known as a SPACK. This comprises two cards, a trunk interface card 30 which is connected to the switch by a T1 link 20, and a base card 35 which is connected to a corresponding adapter card 50 in the RISC System/6000 via link 40 (which is proprietary). The switch 22 also typically includes a card (not shown) to terminate the T1 link 20. Each SPACK can handle 24 telephony channels (ie the number of channels in a single T1 trunk). To support higher numbers of telephony channels, it is necessary to use additional SPACKs (shown schematically as 25A). Each of these has its own T1 link 20A with the switch, and link 40A to a corresponding digital trunk adapter card 50A in the RISC System/6000 computer. Signalling can be exchanged between the switch 22 and the voice processing system 60 either over the T1 link 20, or via a separate connection (not shown).
Note that dual digital trunk adapter cards are available from IBM, which can be used instead of two separate digital trunk adapter cards in the RISC System/6000 (ie two SPACKs can connect to a single dual digital trunk adapter card). The dual digital trunk adapter card fits into the same slot as a single digital trunk adapter card.
In the DirectTalk/6000 system, the SPACKs are provided in one or more digital trunk processor units 80. A single digital trunk processor contains one SPACK and one power supply, whilst a multiple digital trunk processor contains up to five SPACKs plus two power supplies (one is for backup purposes). The DirectTalk/6000 system can process up to 120 telephony channels, corresponding to 5 SPACKs (ie one multiple digital trunk processor). Since a RISC System/6000 computer can only contain three digital trunk adapter cards, it is necessary to use at least two dual digital trunk adapter cards in order to support 5 SPACKs (together with either a single digital trunk adapter card, or another dual digital trunk adapter card).
Thus the telephony connections between the switch and the voice processing system are somewhat complex. Moreover, as computers become increasingly powerful, the number of telephony channels that they can process continues to grow. Thus it is now possible for a voice response unit (VRU), an IVR voice processing system, to handle 10 E1/T1 trunk lines, corresponding to up to 300 telephony channels (for E1). For the configuration of FIG. 1, this would then require 10 cards in the switch and 10 line interface units for the voice processing system, representing a very significant hardware cost. It is potentially possible to reduce costs by the use of dual or quad interface cards, which can interface to two or four T1/E1 trunk lines respectively (similar in concept to the use of the dual digital trunk adapter), but these are still expensive.
Alternatively, it might be possible to use a T3 connection between the switch and the voice processing system. A T3 connection comprises 28 T1 trunks effectively bundled together into a single link, and so provides for a very large number of simultaneous channels between the switch and the voice processing system. However, hardware to support such a T3 interface for a voice processing interface is not commercially available at present, and even if available, such hardware would be very specialised, and hence expensive.
It will be appreciated that many voice processing systems have a somewhat different architecture from that shown in FIG. 1. Typically these are built around an internal time division multiplex (TDM) bus, in which each telephony channel that is being handled by the system is allocated a timeslot on the TDM bus. The main TDM bus types commercially used in voice processing systems are the PCM Expansion bus (PEB), supporting up to 128 64 kbps TDM timeslots, and the SCbus, supporting up to 2048 64 kbps timeslots, both from Dialogic Corporation, and the MVIP bus, defined by the GO-MVIP organisation, and available from Natural Microsystems Corporation and others, supporting up to 512 64 kbps timeslots. In such voice processing systems, the line interface unit is effectively incorporated into the host computer, such that the functionality of the SPACK and the digital trunk adapter of FIG. 1 are essentially combined into a single adapter card. Whilst this approach is simpler in terms of hardware requirements, there are often restrictions on overall call handling capacity, due to physical limitations on the number of adapter cards that can be incorporated into the host computer. Of course, these restrictions can be overcome by using multiple voice processing systems, but this again increases complexity and costs.
Another possibility, since switches themselves are typically built around a TDM bus, is to try to effectively provide a direct extension from the TDM bus of the switch through to the TDM bus of the voice processing system. For example, an SCxbus is available from Dialogic Corporation for bridging between two buses conforming to the above-mentioned SCbus architecture. However, this requires the switch to conform to (or at least be able to interface to) the particular bus architecture of the voice processing system. Moreover, such bus extensions sometimes have limitations on length, thereby restricting the relative physical locations of the switch and the voice processing system.
A somewhat different approach to linking a switch to a voice processing system is disclosed in WO 96/42164. This teaches a system in which the switch is connected to the voice processing system by a standard data connection, such as a SCSI link, an ATM link, etc (see FIG. 7 of this document and associated description). The purpose of this arrangement is to avoid duplicating telephony processing hardware in both the switch and the voice processing line interface unit.
In general, as the demand for voice processing function increases, systems must be able to simultaneously handle increasing numbers of telephony channels. This puts the interface between the switch and an attached voice processing system (or systems) under ever greater strain.